Usage: FFmpeg + TRTC

[winlinvip]

TBD

[wwllww]

Use TRTC with FFmpeg, you must has the sdkAppid and UserSig, The format whip URL like this: https://signal.cloud-rtc.com/v2/pub/sdkappid/roomId?SessionID=userid.

For example: https://signal.cloud-rtc.com/v2/pub/1400704311/610015?SessionID=u85106544

Note: roomId is strRoomId("123” and 123 are not the same room in the TRTC backend service) For TRTC.
sdkAppid must Activate wtn service, otherwise you got error with 403.

To download the code and build FFmpeg, you can use the following command.

cd ~/git 
git clone -b master https://github.com/winlinvip/ffmpeg-webrtc.git
cd ffmpeg-webrtc
./configure --enable-muxer=whip --enable-openssl --enable-version3 \
    --enable-libx264 --enable-gpl --enable-libopus
make -j

Note: To enable DTLS handshake, OpenSSL is mandatory. Please install OpenSSL, for instance, brew install openssl, and then configure the environment by running export PKG_CONFIG_PATH="/usr/local/opt/openssl@3/lib/pkgconfig".

Note: For demonstration purposes, you can install libx264 by running brew install x264 and libopus by running brew install opus.

~/git/ffmpeg-webrtc/ffmpeg -re -f lavfi -i testsrc=size=1280x720 -f lavfi -i sine=frequency=440 -pix_fmt yuv420p \
    -vcodec libx264 -profile:v baseline -r 25 -g 50 -preset:v ultrafast -tune zerolatency -threads 1 \
    -acodec libopus -ar 48000 -ac 2 \
    -handshake_timeout 3000 -pkt_size 1200 \
    -authorization "xxxxxx"  \
    -f whip "https://signal.cloud-rtc.com/v2/pub/1400704311/610015?SessionID=u85106544"

# Or you can also capture your screen, and measure the end-to-end latency.
~/git/ffmpeg-webrtc/ffmpeg -f avfoundation -framerate 25 -pixel_format yuyv422 -i "2:0" \
    -vcodec libx264 -pix_fmt yuv420p -profile:v baseline -preset:v ultrafast \
    -b:v 800k -s 1024x576 -r 25 -g 50 -tune zerolatency -threads 1 -bf 0 \
    -acodec libopus -ar 48000 -ac 2 \
    -handshake_timeout 3000 -pkt_size 1200 \
    -authorization "xxxxxx"  \
    -f whip "https://signal.cloud-rtc.com/v2/pub/1400704311/610015?SessionID=u85106544"

If you want to push mp4 or flv file, you can convert video to H264 baseline and audio convert to Opus.

~/git/ffmpeg-webrtc/ffmpeg -re -i ~/Downloads/test.mp4 -vcodec libx264 \
       -profile:v baseline -preset ultrafast -tune zerolatency -threads 1 -bf 0 \
      -c:a libopus -ar 48000 -ac 2 -ab 32k \
      -handshake_timeout 3000 -pkt_size 1200 \
      -authorization "xxxxxx"  \
      -f whip "https://signal.cloud-rtc.com/v2/pub/1400704311/610015?SessionID=u85106544"

Note:authorization is the usersig generate with sdkAppid and roomId.